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[Freeswitch-users] Accepting SIP Calls from unregistered dev


 
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klaus.teller at gmx.net
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PostPosted: Fri Oct 10, 2008 1:58 pm    Post subject: [Freeswitch-users] Accepting SIP Calls from unregistered dev Reply with quote

Anonynous would be enough for me.

Klaus.
-------- Original-Nachricht --------
Quote:
Datum: Fri, 10 Oct 2008 13:51:21 -0500
Von: Brian West <brian@freeswitch.org>
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Accepting SIP Calls from unregistered devices

Quote:
Anonymous or via IP auth?

/b


On Oct 10, 2008, at 1:48 PM, Klaus Teller wrote:

Quote:
Hi,

How do i configure my Freeswitch to accept SIP calls from peers/
devices not registered with it?

Thanks,
Klaus.
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klaus.teller at gmx.net
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PostPosted: Fri Oct 10, 2008 1:59 pm    Post subject: [Freeswitch-users] Accepting SIP Calls from unregistered dev Reply with quote

Hi,

How do i configure my Freeswitch to accept SIP calls from peers/devices not registered with it?

Thanks,
Klaus.
--
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brian at freeswitch.org
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PostPosted: Fri Oct 10, 2008 1:59 pm    Post subject: [Freeswitch-users] Accepting SIP Calls from unregistered dev Reply with quote

Anonymous or via IP auth?

/b


On Oct 10, 2008, at 1:48 PM, Klaus Teller wrote:

Quote:
Hi,

How do i configure my Freeswitch to accept SIP calls from peers/
devices not registered with it?

Thanks,
Klaus.
--


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brian at freeswitch.org
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PostPosted: Fri Oct 10, 2008 2:17 pm    Post subject: [Freeswitch-users] Accepting SIP Calls from unregistered dev Reply with quote

You have to do this... set auth-calls to false.





<!-- everything above this is public -->
<extension name="check_auth" continue="true">
<condition field="${sip_authorized}" expression="^true$" break="never">
<anti-action application="respond" data="407"/>
</condition>
</extension>
<!-- everything below this requires auth -->

Just wedge this extension in your dialplan between the stuff you want public and the stuff you want auth on. Then make sure you turn auth-calls to false on your profile.


/b





On Oct 10, 2008, at 1:57 PM, Klaus Teller wrote:
Quote:

Anonynous would be enough for me.

Klaus.
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klaus.teller at gmx.net
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PostPosted: Fri Oct 10, 2008 2:31 pm    Post subject: [Freeswitch-users] Accepting SIP Calls from unregistered dev Reply with quote

Hmm, i'm not getting it right. Here is what i did:

1) In profile internal.xml i replaced

<param name="auth-calls" value="true"/> with <param name="auth-calls" value="false"/>

2) In profile external.xml this was already false.

3) Then i added the extension you provided both in public.xml and in default.xml. In public.xml the extension is pasted just before the following line:

<!--
You can place files in the public directory to get included.
-->
<X-PRE-PROCESS cmd="include" data="public/*.xml"/>

In default.xml it is placed just before the similar instructions.

Yet, i still get the following ACL error message from Freeswitch:

IP 192.168.2.34 Rejected by acl domains. My dialpaln and configuration is almost the same i got from trunk one week ago.

I'm sure i'm doing few things wrong here. but what?

Thanks,

Klaus.


-------- Original-Nachricht --------
Quote:
Datum: Fri, 10 Oct 2008 14:01:57 -0500
Von: Brian West <brian@freeswitch.org>
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Accepting SIP Calls from unregistered devices

Quote:
You have to do this... set auth-calls to false.



<!-- everything above this is public -->
<extension name="check_auth" continue="true">
<condition field="${sip_authorized}" expression="^true$"
break="never">
<anti-action application="respond" data="407"/>
</condition>
</extension>
<!-- everything below this requires auth -->

Just wedge this extension in your dialplan between the stuff you want
public and the stuff you want auth on. Then make sure you turn auth-
calls to false on your profile.

/b



On Oct 10, 2008, at 1:57 PM, Klaus Teller wrote:

Quote:

Anonynous would be enough for me.

Klaus.


--
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Jetzt dabei sein: http://www.shortview.de/wasistshortview.php?mc=sv_ext_mf@gmx

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brian at freeswitch.org
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PostPosted: Fri Oct 10, 2008 2:44 pm    Post subject: [Freeswitch-users] Accepting SIP Calls from unregistered dev Reply with quote

Oh yah remove the domains ACL param too it'll trump this Razz

/b

On Oct 10, 2008, at 2:23 PM, Klaus Teller wrote:

Quote:
IP 192.168.2.34 Rejected by acl domains. My dialpaln and
configuration is almost the same i got from trunk one week ago.


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klaus.teller at gmx.net
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PostPosted: Fri Oct 10, 2008 2:55 pm    Post subject: [Freeswitch-users] Accepting SIP Calls from unregistered dev Reply with quote

Great, thanks for the support. I've changed acl.conf.xml in the following way:

<list name="domains" default="deny">
<node type="allow" domain="$${domain}"/>
</list>

into

<list name="domains" default="allow">
<node type="allow" domain="$${domain}"/>
</list>


Regards,

Klaus.

-------- Original-Nachricht --------
Quote:
Datum: Fri, 10 Oct 2008 14:34:33 -0500
Von: Brian West <brian@freeswitch.org>
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Accepting SIP Calls from unregistered devices

Quote:
Oh yah remove the domains ACL param too it'll trump this Razz

/b

On Oct 10, 2008, at 2:23 PM, Klaus Teller wrote:

Quote:
IP 192.168.2.34 Rejected by acl domains. My dialpaln and
configuration is almost the same i got from trunk one week ago.


_______________________________________________
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--
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brian at freeswitch.org
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PostPosted: Fri Oct 10, 2008 3:03 pm    Post subject: [Freeswitch-users] Accepting SIP Calls from unregistered dev Reply with quote

Well removing it off the sofia profile is optimal since you're going
to mix half open half authed.

/b

On Oct 10, 2008, at 2:45 PM, Klaus Teller wrote:

Quote:
Great, thanks for the support. I've changed acl.conf.xml in the
following way:

<list name="domains" default="deny">
<node type="allow" domain="$${domain}"/>
</list>

into

<list name="domains" default="allow">
<node type="allow" domain="$${domain}"/>
</list>


Regards,

Klaus.


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klaus.teller at gmx.net
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PostPosted: Fri Oct 10, 2008 3:50 pm    Post subject: [Freeswitch-users] Accepting SIP Calls from unregistered dev Reply with quote

OK, i moved it to the sofia profile and it works just fine. Next step in my authorization problem. I call in with a softphone and want my call to be transferred via event socket to a remote application.

Yet, Freeswitch rejects the call with the reason:

[407][Proxy Authentication Required] session:sofia/internal/22000@192.168.2.37


I've got the outbound event socket working fine when i call from a peer registered with Freeswitch. But in this new scenario (with a non-registered device calling) Freeswitch is not happy.

Anything i could change to make it work?

Thanks,

Klaus.
-------- Original-Nachricht --------
Quote:
Datum: Fri, 10 Oct 2008 14:47:34 -0500
Von: Brian West <brian@freeswitch.org>
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Accepting SIP Calls from unregistered devices

Quote:
Well removing it off the sofia profile is optimal since you're going
to mix half open half authed.

/b

On Oct 10, 2008, at 2:45 PM, Klaus Teller wrote:

Quote:
Great, thanks for the support. I've changed acl.conf.xml in the
following way:

<list name="domains" default="deny">
<node type="allow" domain="$${domain}"/>
</list>

into

<list name="domains" default="allow">
<node type="allow" domain="$${domain}"/>
</list>


Regards,

Klaus.


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Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

--
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brian at freeswitch.org
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PostPosted: Fri Oct 10, 2008 4:00 pm    Post subject: [Freeswitch-users] Accepting SIP Calls from unregistered dev Reply with quote

Ok I think you and I aren't on the same page... can you clarify what you mean by "my call to be transferred via event socket"?

/b

On Oct 10, 2008, at 3:37 PM, Klaus Teller wrote:
Quote:
OK, i moved it to the sofia profile and it works just fine. Next step in my authorization problem. I call in with a softphone and want my call to be transferred via event socket to a remote application.

Yet, Freeswitch rejects the call with the reason:

[407][Proxy Authentication Required] session:sofia/internal/22000@192.168.2.37 ([email]sofia/internal/22000@192.168.2.37[/email])


I've got the outbound event socket working fine when i call from a peer registered with Freeswitch. But in this new scenario (with a non-registered device calling) Freeswitch is not happy.

Anything i could change to make it work?

Thanks,

Klaus.
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klaus.teller at gmx.net
Guest





PostPosted: Fri Oct 10, 2008 4:05 pm    Post subject: [Freeswitch-users] Accepting SIP Calls from unregistered dev Reply with quote

Well, i want to manage calls remotely using the socket interface (as described here http://wiki.freeswitch.org/wiki/Event_socket_outbound). The calls i want to manage come from non-registered devices.

Does that make sense?

Klaus.
-------- Original-Nachricht --------
Quote:
Datum: Fri, 10 Oct 2008 15:44:09 -0500
Von: Brian West <brian@freeswitch.org>
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Accepting SIP Calls from unregistered devices

Quote:
Ok I think you and I aren't on the same page... can you clarify what
you mean by "my call to be transferred via event socket"?

/b

On Oct 10, 2008, at 3:37 PM, Klaus Teller wrote:

Quote:
OK, i moved it to the sofia profile and it works just fine. Next
step in my authorization problem. I call in with a softphone and
want my call to be transferred via event socket to a remote
application.

Yet, Freeswitch rejects the call with the reason:

[407][Proxy Authentication Required]
session:sofia/internal/22000@192.168.2.37
Quote:


I've got the outbound event socket working fine when i call from a
peer registered with Freeswitch. But in this new scenario (with a
non-registered device calling) Freeswitch is not happy.

Anything i could change to make it work?

Thanks,

Klaus.


--
GMX startet ShortView.de. Hier findest Du Leute mit Deinen Interessen!
Jetzt dabei sein: http://www.shortview.de/wasistshortview.php?mc=sv_ext_mf@gmx

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brian at freeswitch.org
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PostPosted: Fri Oct 10, 2008 4:09 pm    Post subject: [Freeswitch-users] Accepting SIP Calls from unregistered dev Reply with quote

You have to route them to the park app. THen you can command and control them.

/b

On Oct 10, 2008, at 3:56 PM, Klaus Teller wrote:
Quote:
Well, i want to manage calls remotely using the socket interface (as described here [url=http://wiki.freeswitch.org/wiki/Event_socket_outbound)]http://wiki.freeswitch.org/wiki/Event_socket_outbound)[/url]. The calls i want to manage come from non-registered devices.

Does that make sense?

Klaus.
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klaus.teller at gmx.net
Guest





PostPosted: Fri Oct 10, 2008 5:02 pm    Post subject: [Freeswitch-users] Accepting SIP Calls from unregistered dev Reply with quote

There are two things that i think are worth mentioning:

1) when i call from a peer registered to Freeswitch, my remote server is indeed notified and can do different thinks on the call.

2) I can also call registered peers using non-registered peers (thanks to the ACL settings we achieved previously)


But the non-registerepeer cannot call extensions that are redirected to the event socket. I will try praking the call and let you know. I was just wondering why the callf rom registered peers don't need to be parked.

Thanks,

Klaus.
-------- Original-Nachricht --------
Quote:
Datum: Fri, 10 Oct 2008 16:00:18 -0500
Von: Brian West <brian@freeswitch.org>
An: freeswitch-users@lists.freeswitch.org
Betreff: Re: [Freeswitch-users] Accepting SIP Calls from unregistered devices

Quote:
You have to route them to the park app. THen you can command and
control them.

/b

On Oct 10, 2008, at 3:56 PM, Klaus Teller wrote:

Quote:
Well, i want to manage calls remotely using the socket interface (as
described here http://wiki.freeswitch.org/wiki/
Event_socket_outbound). The calls i want to manage come from non-
registered devices.

Does that make sense?

Klaus.


--
Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger

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